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Business-Class Router Solutions – All Telecommunication Services in One Access

4.10 Voice Call Routing


VRT (Voice Routing Table)

www.oneaccess-net.com
v0.11 2007
contact@oneaccess-net.com
Copyright © OneAccess Networks – All rights reserved
VoIP: Call routing

 Overview
 Applicable to any call between any type of ports (local or voip).
 Any call routing is possible except VoIP -> VoIP call
 Implicit Routing
 Easy to configure : no called number configuration
 No number processing
 Routing Table
 Incoming call routing : optional process done for each incoming call (from
VoIP network or local port) allowing number translations (calling & called
numbers) and voice profile selection
 Outgoing call routing : for numbering plan definition (overlap or in-bloc),
selection of the remote voip device if no gatekeeper, number translations
(calling & called)
 Possibility to configure a backup route in case of outgoing call failure
 Possibility to modify specific ISDN Information Elements :
Type of number, Numbering plan, Bearer capability

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Call Routing Architecture (from local port)

Incoming call routing


Outgoing call routing
(number translations)

PSTN Interfaces VoIP Network


(ISDN, analog) (SDP media properties)

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Call Routing Architecture (from VoIP network)

Incoming call routing


(select voip profile,
Outgoing call routing number translations)

PSTN Interfaces VoIP Network


(ISDN, analog) (SDP media properties)

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VoIP: Call Routing

 Up to 200 routing rules can be configured


 Type of rule: incoming (initial routing when receiving the call) or
outgoing (2nd step of routing)
 Number to be checked
 Called or calling number
 Wildcards can be used
 Related port (local or VoIP)
 Actions: Number translations, ISDN I.E. forcing
 Definition of backup routes
 The rule order is important: the rule #1 is checked
first
 Insertion & deletion of rules by using the rule index

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VoIP: Call Routing

 Routing Process for call coming from a local port


 1. The incoming rules related to the local port are checked down the
rule list
 If matching the rule (number & port), the number translations are
applied
 The check of incoming routing rules can be processed again until the
last one. This possibility allows to make several number translations on
the calling and called number.
 2. The outgoing rules are checked from the first one
 If fully matching the rule, the number translations of the rule are applied
and the call is routed to the specified port (local or voip).
 Backup route can be configured to define alternative rules in case
of failure.
 If partially matching the rule (e.g.: incomplete number in overlap mode),
the call is not further routed and the GW waits for another digit

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VoIP: Call Routing

 Routing Process for call coming from VoIP


 1. The incoming rules related to voip are checked from the
first one
 Upon a match (number only), the number translations are applied
 The voip port (coder profile,etc) is determined
 Check of routing table can be processed again until the end of the
table
 2. The outgoing rules are checked from the first one
 Outgoing rules with voip port are excluded (no voip-voip routing)
 Upon a full match, the number translations are applied and the call
is routed to the specified port (local or voip).
 Backup can be configured to define alternative rules in case of failure.
 Upon a partial match (ex : uncomplete number in overlap mode), the
routing process is fully cancelled (done again with the next digit).

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VoIP: Call Routing

 Routing on local port


 Configuration of a port group
 Same number for several ports
 Incoming Calls are distributed with a configurable priority
 Possibility of call hunting
 automatic backup inside the port group
 higher priority is given to « working » ports
 Several numbers can be configured for each port

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Voice Routing (1)

 6. Voice routing

one200(configure)>voice-routing
one200(voice-route)> ?
display-routes - Show voice routing table
exit - Exit from command node
insert - Insert VOIP voice route
move - Move VOIP voice route
no - no
route - Set VOIP voice route
test-route - Test voice routing table
<cr>
one200(voice-route)>

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Voice Routing (2)

one200(voice-route)>route 1
one200(conf-voice-route)> ?
dial-peer - Set route dial peer
exit - Exit from command node
force-bearer-cap - Set force bearer capability fields
force-clir - Set caller line identity request
force-numplan - Set origin/destination numplan
insert-calling - Set route calling insertion
insert-prefix - Set route prefix insertion
insert-suffix - Set route suffix insertion
no - no
prefix - Set route prefix(prefix [number-type][length][timer]
[overlap])
prefix-type - Set route direction and type of call
startup-file - Set restart equipment with the file name.
suppress-prefix - Set route prefix suppression
translate - Set route number translate
wildcard - Set wildcard value for prefix
<cr>

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Voice Routing (3)

 Registration to registrar server (user=phone).

 E164 Prefix is mapped to a pots-group or a voip resource. To register it,


<ua-sip> must be added. Corresponding prefix is used as user portion of
sip uri of To/From/Contact header field

 To register a username instead of a E164 prefix (no user=phone), sip-


username must be validated. It overrides « prefix » value for
To/From/Contact header fields for register method. It overrides « prefix »
value for From/Contact header fields for Invite method.

 Registrar ip address or domain name must be configured at the sip-


gateway (reg-dns-add)

 401 or 407 challenge for register and/or invite method. Sip-


authentication <username> <password> are provided to calculate
MD5 response.

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Voice Routing (4)

 authentication

CLI(voice-route)>route 1
CLI(voice-route)>dial-peer pots-group 0 ua-sip
CLI(voice-route)>sip-username <user portion of sip uri>
CLI(voice-route)>sip-authentication [<username> <password>]
CLI(voice-route)>prefix 1000 length 4
CLI(voice-route)>prefix-type outgoing called last
CLI(voice-route)>exit
CLI(configure)>exit

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Voice Routing (5)

one200(voice-route)>display-routes
1 - 20. all length 4 / voip 1 / mixed - called - next
2 -1000 all length 4 / pots 0 / outgoing - called
- last

one200>show running-config
...
voice-routing
route 1
dial-peer voip 1
prefix 20. length 4
prefix-type mixed called next
exit
route 2
dial-peer pots-group 1 ua-sip
prefix 1000 length 4
prefix-type outgoing called last
exit
exit
...

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Voice Routing (6)

 Prefix suppression

1234

Pots-group 0 12011234

VoIP 0

route 1
dial-peer pots-group 0
prefix-type outgoing called last
prefix 120A. length 4
suppress-prefix 4 called
wildcard A 0123
exit

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Voice Routing (7)

 Codec Profile for incoming call

Pots-group 0 Incoming 986902739


VoIP 1 Table

VoIP 2
Pots-group 4 986911111

route 14
dial-peer voip 1
prefix 986902739 length 9
prefix-type incoming called
exit
route 15
dial-peer voip 2
prefix . length 0
prefix-type incoming called
exit

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Voice Routing (8)

 Modification of Calling and Called numbers 1/4

Calling : 019869027391111 Incoming Calling : 986902739 Outgoing


Called : 12345678 Table Called : 120012345678 Table

Calling : 986902735
Calling : 019869027352222 Called : 12345678
Called : 12345678

Pots-group 0

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Voice Routing (9)

 Modification of Calling and Called numbers 2/4

route 2
dial-peer pots-group 0
translate .98690273A. 98690273A calling
prefix . length 0
prefix-type incoming calling next
exit

route 3
dial-peer pots-group 0 alias
insert-prefix 1200 called
prefix 986902739 length 9
prefix-type incoming calling
exit

alias added to register the number 1200 in the RAS

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Voice Routing (10)

 Modification of Calling and Called numbers 3/4

Calling : 986902739
Called : 120012345678
Calling : 019869027391111 Incoming Calling : 986902739 Outgoing
Called : 12345678 Table Called : 120012345678 Table

Voip 1
Calling : 986902735
Calling : 019869027352222 Called : 12345678
Called : 12345678 Voip 0

Pots-group 0

Calling : 986902735
Called : 12345678

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Voice Routing (11)

 Modification of Calling and Called numbers 4/4


route 4
dial-peer voip 1
prefix 1200. length 4
prefix-type outgoing called last
exit

route 5
dial-peer voip 0
prefix . length 0
prefix-type outgoing called last
exit

Copyright © OneAccess Networks – All rights reserved 19


Business-Class Router Solutions – All Telecommunication Services in One Access

Voice Toubleshooting & Statistics

www.oneaccess-net.com
v0.11 2007
contact@oneaccess-net.com
Copyright © OneAccess Networks – All rights reserved
Voice routing table testing

One_Training(voice-route)>display-route
1 - . all length 10 / pots 0 / incoming - called - last
/ +prefix-calling 0
10 - . all timer / voip 0 / outgoing - called - last
20 - . all length 9 / pots 0 / outgoing - called - last
/ -prefix-called 1
One_Training(voice-route)>

One_Training> configure terminal


One_Training(configure)>voice-routing
One_Training(voice-route)>test-route 0141877422 260402020 from-voip 0
--> test_route : Calling=0141877422 Called=260402020
Route match : Calling=0141877422 Called=60402020
Incoming Routes = Outgoing Routes = 20,
Incoming call from voip id:0 Send towards Isdn on local port: 5/0
One_Training(voice-route)>
One_Training(voice-route)>
One_Training(voice-route)>test-route 260402020 0141877422 from-pots 0
--> test_route : Calling=260402020 Called=0141877422
Route match : Calling=0260402020 Called=0141877422
Incoming Routes = 1, Outgoing Routes = 10,
Outgoing call from local port: 5/0 -> Send towards H323 on voip id:0
One_Training(voice-route)>

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VoIP Statistics

 BRI statistics
CLI# show voice voice-port bri index 0
voice port 5/0
protocol descriptor BRI_NT
current state activated
config state up
layer 1 status activated
attached vmoabri dial peer 0
number of voice communication 0
bri Tx frames on D channel 40
bri Rx frames on D channel 41

Outgoing calls : 102


Outgoing calls failures : 5
Physical Interface down : 0
Cause Class 0 (normal event) : 0
Cause Class 1 (normal event) : 5
Normal Cause (16) : 2
User busy (17) : 3
No answer (18) : 0
Cause Class 2 (unavailable ressources) : 0
Cause Class 3 (unavailable service) : 0
Cause Class 4 (service not provided) : 0
Cause Class 5 (invalid message) : 0
Cause Class 6 (protocol error) : 0
Cause Class 7 (interworking) : 0

Incoming calls : 54
Incoming calls failures : 7
Remote failure : 0
Unknown number : 5
DSP unavailable : 0
Not specified : 2

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VoIP Statistics

 PRI statistics
CLI# show voice voice-port pri index 0
voice port 5/0
physical type E1
protocol descriptor E1_PRI
current state activated
config state up
layer 1 status deactivated
number of voice communications 0
pri AIS occurence 0
pri RDI occurence 0

Outgoing calls : 67
Outgoing calls failures : 3
Physical Interface down : 0
Cause Class 0 (normal event) : 0
Cause Class 1 (normal event) : 3
Normal Cause (16) : 0
User busy (17) : 3
No answer (18) : 0
Cause Class 2 (unavailable ressources) : 0
Cause Class 3 (unavailable service) : 0
Cause Class 4 (service not provided) : 0
Cause Class 5 (invalid message) : 0
Cause Class 6 (protocol error) : 0
Cause Class 7 (interworking) : 0

Incoming calls : 23
Incoming calls failures : 2
Remote failure : 2
Unknown number : 0
DSP unavailable : 0
Not specified : 0

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VoIP Statistics

 FXS statistics
CLI# show voice voice-port fxs index 0
voice port 5/0
current state on hook
config state up
attached vmoa fxs dial peer 0
voice communication no

Outgoing calls : 32
Outgoing calls failures : 3
User busy : 2
No answer : 1
Incoming calls : 6
Incoming calls failures : 0
Remote failure : 0
Unknown number : 0
DSP unavailable : 0
Not specified : 0

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VoIP Statistics

 Dial-peer VoIP Statistics (1)


one200> show voice dial-peer voice voip type index <port id> [reset]
or
one200> show voice dial-peer voice voip type global[reset]

type may be :
-current : statistics on current calls
-outgoing : outgoing calls only
-incoming : incoming calls only
-user-plan : voice & fax only
-all (default) : all the statistics are provided

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VoIP Statistics

 Dial-peer VoIP Statistics (2): Outgoing Calls


Dial Peer 1
Current Calls 0
Outgoing Calls
Outgoing Calls 4
Outgoing calls failures 0
RAS Call Failures 0
Gatekeeper Unavailable 0
Admission Rejects 0
H225/Q931 Call failures 0
Cause Class 0 (normal event) 0
Cause Class 1 (normal event) 0
Normal Cause (16) 0
User busy (17) 0
No answer (18) 0
Cause Class 2 (unavailable ressources) 0
Cause Class 3 (unavailable service) 0
Cause Class 4 (service not provided) 0
Cause Class 5 (invalid message) 0
Cause Class 6 (protocol error) 0
Cause Class 7 (interworking) 0
H245 Call failures 0
Incompatible capabilities 0
Protocol errors 0
Internal call failures 0
DSP unavailable 0
Max-bandwidth exceeded 0
Max-connection exceeded 0
Not specified 0

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VoIP Statistics

 Dial-peer VoIP Statistics (3): Incoming Calls


Incoming calls 3
Incoming calls failures 1
RAS Call failures 0
Gatekeeper Unavailable 0
Admission Rejects 0
Local Port Call failures 1
H245 Call failures 0
Incompatible capabilities 0
Protocol errors 0
Internal call failures 0
DSP unavailable 0
Unknown number 0
Channel / port unavailable 0
Max-bandwidth exceeded 0
Max-connection exceeded 0
Not specified 0

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VoIP Statistics

 Dial-peer VoIP Statistics (4): Voice and Fax


RTP statistics
Number of transmitted packets 1237
Number of received packets 1234
Number of transmitted bytes 101484
Number of received bytes 101098
Number of excessive jitter events 3
Number of lost packets 0
Number of invalid packets 0
Number of calls with frame error rate
total <0.01<0.1<0.5<1<5>=5
3 0 0 2 1 0 0
Modem passthrough
Number of switching to modem mode 0
T38 FAX Calls
Number of outgoing fax 0
Number of incoming fax 0
Number of failures 0
Request Mode failure 0
Pre-message procedure failure 0
Page failure 0
Number of transmitted packets 0
Number of received packets 0
Number of transmitted bytes 0
Number of received bytes 0
Number of lost packets 0
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VoIP Statistics

 Events
vxTarget>event
filter - Add/remove events filters
manager - Add a SNMP manager
no - No
recover - Recover events from memory
vxTarget>event filter
add - Add an event filter
remove - Remove a events filter from the table
vxTarget>event filter add vox
ALL - All families from vox group
GEN - GEN
VOATM - VOATM
VOIP - VOIP
vxTarget>event filter add vox voip
<subfam> - <ALL | ControlPlan | UserPlan>
<fam2> - <GEN | VOATM>
vxTarget>event filter add vox voip all show

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VoIP Statistics

 Voice call history, active calls


 Gives statistics on the current voice calls and the last 100
calls

vxTarget>show voice voip-call any ind 1

1 - Call from remote voip: 0, to local port: 5/1 call-id: 4 active


calling : 110, called : 111
setup time: 01/02/00 04h58m31s

01/02/00 04h58m31s
RTP Source ip :192.168.1.110 rtp:16384 /Dest ip :192.168.1.111 rtp:16386 (active)
Play time (voice) : 00h00m39s
Tx Coder : G729 / 20 ms ; Rx Coder : G729
RTP Packets RX / TX : 1988 / 1989
RTP Packet lost&discarded RX / TX (RTCP reported) : 0 / 395307
Number of Excessive Jitter events : 3

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VoIP Statistics

 RTP sessions history


 Gives complete statistics about the 200 last RTP sessions

CLI> show voice rtpcall full any ind 2

2 - 01/04/01 00h47m24s RTP 192.168.1.1:16384 – 192.168.1.111:16386


Play time (voice) : 00h00m46s
Tx Coder : G729 / 20 ms ; Rx Coder : G729
VAD enabled local / remote : no / no

ERL : 15 dB
ACOM : 32 dB

RTP Packets received (DSP / Uplink) : 2337 / 2337


lost : 0
out of sequence : 0
invalid : 0
RTP Packets transmitted (DSP / Uplink) : 2338 / 2338
lost (RTCP reported) : 0

Jitter parameter : 100 ms


Number of Excessive Jitter events : 1

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VoIP Statistics

 RTP sessions history (continue)


Excessive Jitter events :
2|
1| *
----------------------------------------
0 30" 1' 2' 4' 8' 12' >16'

Jitter received (uplink) :


Max delay : 93 ms
Delays (ms) >50 >100 >150 >200 >300
Nb of occur. 2 0 0 0 0
Interarrival max jitter : 9 ms
Jitter received (DSP) :
Max delay : 93 ms
Delays (ms) >50 >100 >150 >200 >300
Nb of occur. 2 0 0 0 0
Interarrival max jitter : 9 ms

Frames with a delay >50 ms :


2|
1| * *
----------------------------------------
0 30" 1' 2' 4' 8' 12' >16'

Jitter transmitted (uplink) :


Max delay : 6 ms
Delays (ms) >50 >100 >150 >200 >300
Nb of occur. 0 0 0 0 0
Interarrival max jitter : 1 ms
(RTCP reported) : 2 ms
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VoIP: Internal Call Generator

 Possibility to generate and / or terminate one or


several VoIP calls
 Two services: RTP loopback or BERT testing
 Use of virtual and routable dial-peer pots
dial-peer voice pots 4
service bert2047 both 3
pots-group 0
exit

voip-call 1
pots 4
called 0198723000
calling 3000
bearer data
duration 180
timeout 10
exit

One200> start identifier 1

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VoIP: ISDN capture

 Possibility to capture the signalling traffic over


ISDN BRI & PRI interfaces: layer 1 to 3
 For VoIP side: use of the IP capture possibilities
CLI>conf t
CLI>logging buffered debug
CLI>exit
CLI>debug isdn all layer 1to3

00:07:21.271 line:5/0 L1 frame sent.


00:07:21.271 line:5/0 L2 tx UI P/F=0 NR=4 NS=2 C/R=1.
00:07:21.271 hex: 02 ff 03
00:07:21.271 line:5/0 L3 tx SETUP callref:8.
00:07:21.271 hex1: 08 01 08 05 04 03 80 90 a3 18 01 89 70 04 a1 31
00:07:21.272 hex2: 31 31 a1
00:07:21.343 line:5/0 L1 frame received.
00:07:21.343 line:5/0 L2 rx SABME P/F=1 C/R=0.
00:07:21.343 hex: 00 01 7f
00:07:21.343 line:5/0 L1 frame sent.
00:07:21.343 line:5/0 L2 tx UA P/F=1 NR=4 NS=2 C/R=0.
00:07:21.343 hex: 00 01 73

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Call factory over IP

 For debug, a SETUP can be sent on VoIP.


One_training>auto-call
<called> - called number: up to 21 characters <0..9, #, *>
One_training>auto-call 0141877422
<calling> - calling number: up to 21 characters <0..9, #, *>
<pots-number> - pots: 0..29
<bearer> - bearer capability < voice | data | voiceband >
overlap - units in milliseconds: 0..2000 <0 means no overlap used>
<cr>
One_training>auto-call 0141877422 2408882005
17:50:17.677 Info vox factory test 1 call-id: 4, ident: auto-call, CALL IN PROGRESS Calling=2408882
005 Called=0141877422.
one100_interopBW>17:50:17.678 Info vox voip controlplan 3 Incoming call on local pots: 0, calling:
2408882005, called: 0141877422, call-id: 4.
17:50:17.710 Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 2408882005, called:
0141877422, call-id: 4.
17:50:27.660 Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED cause=no codec.
17:50:27.661 Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED on pots cause=[Norma
l call clearing].

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Auto call to ISDN

 For debug, a ‘SETUP’ can be sent on a ISDN local


port. One_training>isdn test call 85841 ( data call/unrestricted )
02:27:34.914 line:5/0 L1 event received PH_AR State:F3.
02:27:34.923 line:5/0 L1 event received EV_LOST_FRAMING State:F4.
02:27:34.923 line:5/0 L1 event received EV_INFO_2 State:F5.
02:27:34.925 line:5/0 L1 event received EV_INFO_4_8(PH_AI) State:F6.
02:27:34.925 line:5/0 L1 event received MPH_AI State:F7.
02:27:34.925 line:5/0 L1 frame sent.
02:27:34.925 line:5/0 L2 tx SABME P/F=1 C/R=0.
02:27:34.925 hex: 00 83 7f
02:27:34.938 line:5/0 L1 frame received.
02:27:34.938 line:5/0 L2 rx UA P/F=1 C/R=0.
02:27:34.939 hex: 00 83 73
02:27:34.939 line:5/0 L1 frame sent.
02:27:34.940 line:5/0 L2 tx INFO P=0 NR=0 NS=0 C/R=0.
02:27:34.940 hex: 00 83 00 00
02:27:34.940 line:5/0 L3 tx SETUP callref:4.
02:27:34.940 hex1: 08 01 04 05 04 02 88 90 18 01 83 70 06 81 38 35
02:27:34.940 hex2: 38 34 31 a1

02:27:34.940 Called Number : 85841

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Fax relay T38 debug

 Fax T38 processing and traces: display of the T30


messages.
One_training>trace filter add vox up ifp 2 show
01:42:38.100 Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 110,
called: 111, call-id: 29.
01:42:40.397 Info vox voip controlplan 3 Alert in band received, call-id: 29.
01:42:40.917 Info vox voip userplan 3 VoIP RTP transmission started, coder: G729, call-id: 29.
01:42:40.997 Info vox voip userplan 3 VoIP RTP reception started, coder: G729, call-id: 29.
01:42:52.444 Info vox voip controlplan 3 Call connected, call-id: 29.
01:42:57.537 Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G729, call-id: 29.
01:42:57.537 Info vox voip userplan 3 VoIP RTP reception stopped, coder: G729, call-id: 29.
01:42:57.665 Info vox voip userplan 3 Fax T38 starting call-id: 29 .
01:43:04.825 VOX.up.ifp.2 ------ NSF / CSI / DIS ------>
01:43:07.717 VOX.up.ifp.2 <-------- TSI / DCS ----------
01:43:07.837 VOX.up.ifp.2 <------- TCF (v29_9600) --------
01:43:11.814 Info vox voip userplan 1 T38 Pre-message procedure OK, call-id: 29.
01:43:11.814 VOX.up.ifp.2 ------------ CFR ------------>
01:43:12.437 Info vox voip userplan 1 T38 Transmitting page 1, call-id: 29.
01:43:12.437 VOX.up.ifp.2 <------ PAGE (v29_9600) --------
01:51:10.502 VOX.up.ifp.2 <--------- PPS-EOP -----------
01:51:13.599 Info vox voip userplan 1 T38 page 1 OK, call-id: 29.
01:51:13.599 VOX.up.ifp.2 ------------ MCF ------------>
01:51:15.282 VOX.up.ifp.2 <----------- DCN -------------
01:51:15.846 Info vox voip controlplan 3 Call Disconnection received on local port: 5/2,
cause: (16)[Normal call clearing], call-id: 29.
Copyright © OneAccess Networks – All rights reserved 37
Modem / fax pass-through event

 Example of event in case of modem/fax call.

One_training>event filter add vox all show

00:11:21.485 Info vox voip controlplan 3 Incoming call on voip id: 0, calling: 111,
called: 110, call-id: 3.
00:11:21.486 Info vox voip controlplan 3 Outgoing call on local port: 5/0, calling: 111,
called: 110, call-id:3.
00:11:21.550 Info vox voip controlplan 3 Alert received, call-id: 3.
00:11:21.631 Info vox voip controlplan 3 Call connected, call-id: 3.
00:11:25.142 Info vox voip userplan 1 Fax/Modem Passthrough starting call-id: 3.
00:11:25.142 Info vox voip userplan 3 RTP new transmission coder: G711 A Law, call-id: 3.
00:11:25.341 Info vox voip userplan 3 RTP new reception coder: G711 A Law, call-id: 3.
00:12:23.533 Info vox voip controlplan 3 Call Disconnection received on local port: 5/1,
cause: (16)[Normal call clearing], call-id: 3.
00:12:23.552 Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G711 A Law, call-id: 3.
00:12:23.552 Info vox voip userplan 3 VoIP RTP reception stopped, coder: G711 A Law, call-id: 3.

Copyright © OneAccess Networks – All rights reserved 38

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